Re: [asterisk-users] quickstart
Am 17.06.2014 17:36, schrieb thufir: On Tue, 17 Jun 2014 12:14:05 +0200, Rainer Piper wrote: git clone https://github.com/asterisk/pjproject pjproject At the very least, thank you for pjsip. I'm not sure what it is yet, but seems intriguing :) I'm on Ubunutu 14.04, but will look over your script and adapt it. !!! take a look at the install_prereq script. !!! you have to install same dependency libs before you compile asterisk !!! and install_prereq just supports ... debian, redhat and OpenBSD # The distributions we do support: if [ -r /etc/debian_version ]; then handle_debian elif [ -r /etc/redhat-release ]; then handle_rh elif [ "$OS" = 'OpenBSD' ]; then handle_obsd fi -Thufir -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] quickstart
On Tue, 17 Jun 2014 09:46:22 -0500, Rusty Newton wrote: > Try following: https://wiki.asterisk.org/wiki/display/AST/Hello+World > > Simply use a hard phone instead of a soft-phone. Then go from there on > to two phones. Perfect. I'm looking at: "SIP channel driver you wanted to use, which may imply other requirements. For example if you want to use chan_pjsip..." so now know to read up, in particular, on what sip channels are :) The book is just so huge, it's hard to find somewhere to start, and this looks like good place. Thank you, everyone, for the responses, I'm off to the races now. -Thufir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] quickstart
On Tue, 17 Jun 2014 12:14:05 +0200, Rainer Piper wrote: > git clone https://github.com/asterisk/pjproject pjproject At the very least, thank you for pjsip. I'm not sure what it is yet, but seems intriguing :) I'm on Ubunutu 14.04, but will look over your script and adapt it. -Thufir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF transmitting letter A
A is a valid DTMF "digit", chances are your PBX is detecting the digit wrong. If you have relaxdtmf enabled, disable it. If that doesn't help, play with the audio gains. Too loud or too soft can cause DTMF issues. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Markus Sent: Tuesday, June 17, 2014 11:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] DTMF transmitting letter A Dear list, maybe not really an Asterisk question, but... all my users dial in via PSTN (via SIP DIDs) and enter a target number via DTMF through my Asterisk 1.4. Out of about 150,000 calls per month I see on average about 1 call per month where an arbitrary caller enters the letter 'A' via DTMF. These callers use their mobile phones to dial in. I just reread the Wikipedia article on DTMF but I don't understand how someone can send an 'A'. Any clue? Thank you! Markus -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF transmitting letter A
Dear list, maybe not really an Asterisk question, but... all my users dial in via PSTN (via SIP DIDs) and enter a target number via DTMF through my Asterisk 1.4. Out of about 150,000 calls per month I see on average about 1 call per month where an arbitrary caller enters the letter 'A' via DTMF. These callers use their mobile phones to dial in. I just reread the Wikipedia article on DTMF but I don't understand how someone can send an 'A'. Any clue? Thank you! Markus -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] quickstart
The headphones are Cisco phones. Ie, ext 100 and 101. I don't have the model handy at the moment. On Jun 17, 2014 2:10 PM, "binary dreamer" wrote: > hi, sorry all you are asking is to have 2 internal phones call each other? > the hardphones you are talking about what kind of phones are? > > > > On Tue, Jun 17, 2014 at 1:14 PM, Rainer Piper > wrote: > >> Am 17.06.2014 09:04, schrieb thufir: >> >> I have the Asterisk book, it's enormous, the 4th edition as per >> http://www.asteriskdocs.org/. >> >> I'd like to do something like: >> >> http://www.voip-info.org/wiki/view/Asterisk+quickstart >> >> just to have two hardphones act as extensions and call each other. Is >> that a reasonable first task? >> >> I'm looking for the simplest litmus test for functionality possible. >> >> >> >> thanks, >> >> Thufir >> >> Hi ... this script will get you up and running on a debian7 >> distribution. >> >> >> #!/bin/sh >> >> apt-get update && apt-get upgrade -y >> >> asteriskversion=asterisk-12.3.2 >> >> apt-get install -y linux-headers-`uname -r` >> apt-get install -y build-essential >> apt-get install -y wget >> apt-get install -y libssl-dev >> apt-get install -y libncurses5-dev >> apt-get install -y libnewt-dev >> apt-get install -y libxml2-dev >> apt-get install -y libsqlite3-dev >> apt-get install -y libjansson-dev >> apt-get install -y git >> >> ln -s /usr/src/linux-headers-`uname -r` /usr/src/linux >> >> cd /usr/src >> >> ## pjsip installieren >> git clone https://github.com/asterisk/pjproject pjproject >> cd /usr/src/pjproject >> ./configure --prefix=/usr --enable-shared --disable-sound >> --disable-resample --disable-video --disable-opencore-amr >> >> ## um IPv6 Support in pjsip einzuschalten, muss das >> CFLAGS='-DPJ_HAS_IPV6=1' angegeben werden >> # IPV6 is turned off at default ! >> #./configure CFLAGS='-DPJ_HAS_IPV6=1' --prefix=/usr --enable-shared >> --disable-sound --disable-resample --disable-video --disable-opencore-amr >> # >> >> make dep >> make >> make install >> ldconfig >> >> ### check inst. >> # ldconfig -p | grep libpj >> >> ## System vorbereiten >> ## download Asterisk >> if [ ! -f /usr/src/$asteriskversion.tar.gz ] ; then >> wget >> http://downloads.asterisk.org/pub/telephony/asterisk/$asteriskversion.tar.gz >> fi >> if [ ! -d /usr/src/$asteriskversion ] ; then >> tar xvzf $asteriskversion.tar.gz >> fi >> ## erforderliche libs installieren >> /usr/src/$asteriskversion/contrib/scripts/install_prereq install >> >> ## optional >> /usr/src/$asteriskversion/contrib/scripts/get_mp3_source.sh >> /usr/src/$asteriskversion/contrib/scripts/get_ilbc_source.sh >> gcc -O2 /usr/src/$asteriskversion/contrib/utils/rawplayer.c -o >> /usr/bin/rawplayer >> >> ## asterisk installieren >> cd /usr/src/$asteriskversion >> ./configure >> make menuconfig >> make >> make install >> make samples >> make config >> make install-logrotate >> >> >> >> >> -- >> *Rainer Piper* >> Integration engineer >> Koeslinstr. 56 >> 53123 BONN >> GERMANY >> Phone: +49 228 97167161 >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >>http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] quickstart
Pardon. My home PC is Ubuntu, 14.04. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] quickstart
On Tue, Jun 17, 2014 at 2:04 AM, thufir wrote: > I have the Asterisk book, it's enormous, the 4th edition as per > http://www.asteriskdocs.org/. > > I'd like to do something like: > > http://www.voip-info.org/wiki/view/Asterisk+quickstart > > just to have two hardphones act as extensions and call each other. Is that a > reasonable first task? > > I'm looking for the simplest litmus test for functionality possible. Once you install asterisk: https://wiki.asterisk.org/wiki/display/AST/Installing+Asterisk Try following: https://wiki.asterisk.org/wiki/display/AST/Hello+World Simply use a hard phone instead of a soft-phone. Then go from there on to two phones. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com & http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Request for spandsp paid support
On Tue, Jun 17, 2014 at 8:26 AM, Ramesh Hegde wrote: > Hello > Does anyone know if there is anyone/any company which does paid support for > spandsp? We are looking for such a company/individual who will support > spandsp based on on specific Service level agrements > > Regards > Ramesh Hi! Please use the asterisk-biz mailing list for commercial/paid support/job posting type discussions. http://lists.digium.com/mailman/listinfo/asterisk-biz -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com & http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Request for spandsp paid support
Hello Does anyone know if there is anyone/any company which does paid support for spandsp? We are looking for such a company/individual who will support spandsp based on on specific Service level agrements Regards Ramesh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - How to configure Apache2 server to receive Polycom log files ?
Hi,
I have this configuration in apache site configuration
RewriteEngine On
RewriteCond %{REQUEST_METHOD} =PUT [OR]
RewriteCond %{REQUEST_METHOD} =HEAD
RewriteRule ^(.*)$ put.php?url=$1
this redirect PUT method to put.php script which read input and write to
file in logs directory.
and this is file put.php:
BR
Stepan
Dne 16.6.2014 15:51, Olivier napsal(a):
Hello,
To troubleshoot Polycom phone provisionning (with an asterisk 11 box),
I'm looking to enable HTTP log file upload ie the capability for
Polycom phones to upload some data to a given HTTP server.
At the moment, Polycom phones are downloading config files from an
Apache2 HTTP server, thanks to a DHCP server configuration option bellow.
option tftp-server-name "http://192.168.64.250/polycom";;
Looking at Apache2 log files, I can see that Polycom phones are trying
to upload log files but every attempt to upload data (fails with 405
error, no matter how I configured target upload directory ownership. See:
192.168.64.215 - - [16/Jun/2014:15:25:03 +0200] "PUT
/polycom/log/0004f2394356-boot.log HTTP/1.1" 405 582 "-"
"FileTransport PolycomSoundPointIP-SPIP_650-UA/4.3.0.0246
"
Has someone successfully received Polycom file uploads with an HTTP
server (ie without using FTP) or is it something can't simply be done ?
If positive, can you share key configuration settings ?
Regards
--
S pozdravem / with kind regards
S(te(pán Hradský
odde(lení specialistu* hlasových sluz(eb / voice department
ha-vel internet s.r.o.
Oles(ní 587/11A
712 00 Ostrava Muglinov
Czech Republic
T +420 552 305 370/ F +420 552 305 306
Hotline: +420 552 305 321
http://www.ha-vel.cz
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
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http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] quickstart
what is your Linux box (debian, Ubuntu, centos, ...)? On Tue, Jun 17, 2014 at 2:10 PM, binary dreamer wrote: > hi, sorry all you are asking is to have 2 internal phones call each other? > the hardphones you are talking about what kind of phones are? > > > > On Tue, Jun 17, 2014 at 1:14 PM, Rainer Piper > wrote: > >> Am 17.06.2014 09:04, schrieb thufir: >> >> I have the Asterisk book, it's enormous, the 4th edition as per >> http://www.asteriskdocs.org/. >> >> I'd like to do something like: >> >> http://www.voip-info.org/wiki/view/Asterisk+quickstart >> >> just to have two hardphones act as extensions and call each other. Is >> that a reasonable first task? >> >> I'm looking for the simplest litmus test for functionality possible. >> >> >> >> thanks, >> >> Thufir >> >> Hi ... this script will get you up and running on a debian7 >> distribution. >> >> >> #!/bin/sh >> >> apt-get update && apt-get upgrade -y >> >> asteriskversion=asterisk-12.3.2 >> >> apt-get install -y linux-headers-`uname -r` >> apt-get install -y build-essential >> apt-get install -y wget >> apt-get install -y libssl-dev >> apt-get install -y libncurses5-dev >> apt-get install -y libnewt-dev >> apt-get install -y libxml2-dev >> apt-get install -y libsqlite3-dev >> apt-get install -y libjansson-dev >> apt-get install -y git >> >> ln -s /usr/src/linux-headers-`uname -r` /usr/src/linux >> >> cd /usr/src >> >> ## pjsip installieren >> git clone https://github.com/asterisk/pjproject pjproject >> cd /usr/src/pjproject >> ./configure --prefix=/usr --enable-shared --disable-sound >> --disable-resample --disable-video --disable-opencore-amr >> >> ## um IPv6 Support in pjsip einzuschalten, muss das >> CFLAGS='-DPJ_HAS_IPV6=1' angegeben werden >> # IPV6 is turned off at default ! >> #./configure CFLAGS='-DPJ_HAS_IPV6=1' --prefix=/usr --enable-shared >> --disable-sound --disable-resample --disable-video --disable-opencore-amr >> # >> >> make dep >> make >> make install >> ldconfig >> >> ### check inst. >> # ldconfig -p | grep libpj >> >> ## System vorbereiten >> ## download Asterisk >> if [ ! -f /usr/src/$asteriskversion.tar.gz ] ; then >> wget >> http://downloads.asterisk.org/pub/telephony/asterisk/$asteriskversion.tar.gz >> fi >> if [ ! -d /usr/src/$asteriskversion ] ; then >> tar xvzf $asteriskversion.tar.gz >> fi >> ## erforderliche libs installieren >> /usr/src/$asteriskversion/contrib/scripts/install_prereq install >> >> ## optional >> /usr/src/$asteriskversion/contrib/scripts/get_mp3_source.sh >> /usr/src/$asteriskversion/contrib/scripts/get_ilbc_source.sh >> gcc -O2 /usr/src/$asteriskversion/contrib/utils/rawplayer.c -o >> /usr/bin/rawplayer >> >> ## asterisk installieren >> cd /usr/src/$asteriskversion >> ./configure >> make menuconfig >> make >> make install >> make samples >> make config >> make install-logrotate >> >> >> >> >> -- >> *Rainer Piper* >> Integration engineer >> Koeslinstr. 56 >> 53123 BONN >> GERMANY >> Phone: +49 228 97167161 >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >>http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users >> > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] quickstart
hi, sorry all you are asking is to have 2 internal phones call each other? the hardphones you are talking about what kind of phones are? On Tue, Jun 17, 2014 at 1:14 PM, Rainer Piper wrote: > Am 17.06.2014 09:04, schrieb thufir: > > I have the Asterisk book, it's enormous, the 4th edition as per > http://www.asteriskdocs.org/. > > I'd like to do something like: > > http://www.voip-info.org/wiki/view/Asterisk+quickstart > > just to have two hardphones act as extensions and call each other. Is that > a reasonable first task? > > I'm looking for the simplest litmus test for functionality possible. > > > > thanks, > > Thufir > > Hi ... this script will get you up and running on a debian7 distribution. > > > #!/bin/sh > > apt-get update && apt-get upgrade -y > > asteriskversion=asterisk-12.3.2 > > apt-get install -y linux-headers-`uname -r` > apt-get install -y build-essential > apt-get install -y wget > apt-get install -y libssl-dev > apt-get install -y libncurses5-dev > apt-get install -y libnewt-dev > apt-get install -y libxml2-dev > apt-get install -y libsqlite3-dev > apt-get install -y libjansson-dev > apt-get install -y git > > ln -s /usr/src/linux-headers-`uname -r` /usr/src/linux > > cd /usr/src > > ## pjsip installieren > git clone https://github.com/asterisk/pjproject pjproject > cd /usr/src/pjproject > ./configure --prefix=/usr --enable-shared --disable-sound > --disable-resample --disable-video --disable-opencore-amr > > ## um IPv6 Support in pjsip einzuschalten, muss das > CFLAGS='-DPJ_HAS_IPV6=1' angegeben werden > # IPV6 is turned off at default ! > #./configure CFLAGS='-DPJ_HAS_IPV6=1' --prefix=/usr --enable-shared > --disable-sound --disable-resample --disable-video --disable-opencore-amr > # > > make dep > make > make install > ldconfig > > ### check inst. > # ldconfig -p | grep libpj > > ## System vorbereiten > ## download Asterisk > if [ ! -f /usr/src/$asteriskversion.tar.gz ] ; then > wget > http://downloads.asterisk.org/pub/telephony/asterisk/$asteriskversion.tar.gz > fi > if [ ! -d /usr/src/$asteriskversion ] ; then > tar xvzf $asteriskversion.tar.gz > fi > ## erforderliche libs installieren > /usr/src/$asteriskversion/contrib/scripts/install_prereq install > > ## optional > /usr/src/$asteriskversion/contrib/scripts/get_mp3_source.sh > /usr/src/$asteriskversion/contrib/scripts/get_ilbc_source.sh > gcc -O2 /usr/src/$asteriskversion/contrib/utils/rawplayer.c -o > /usr/bin/rawplayer > > ## asterisk installieren > cd /usr/src/$asteriskversion > ./configure > make menuconfig > make > make install > make samples > make config > make install-logrotate > > > > > -- > *Rainer Piper* > Integration engineer > Koeslinstr. 56 > 53123 BONN > GERMANY > Phone: +49 228 97167161 > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] quickstart
Am 17.06.2014 09:04, schrieb thufir: I have the Asterisk book, it's enormous, the 4th edition as per http://www.asteriskdocs.org/. I'd like to do something like: http://www.voip-info.org/wiki/view/Asterisk+quickstart just to have two hardphones act as extensions and call each other. Is that a reasonable first task? I'm looking for the simplest litmus test for functionality possible. thanks, Thufir Hi ... this script will get you up and running on a debian7 distribution. |#!/bin/sh| |apt-get update && apt-get upgrade -y| |asteriskversion=asterisk-12.3.2| | apt-get install -y linux-headers-`uname -r` apt-get install -y build-essential apt-get install -y wget apt-get install -y libssl-dev apt-get install -y libncurses5-dev apt-get install -y libnewt-dev apt-get install -y libxml2-dev apt-get install -y libsqlite3-dev apt-get install -y libjansson-dev apt-get install -y git ln -s /usr/src/linux-headers-`uname -r` /usr/src/linux cd /usr/src ## pjsip installieren git clone https://github.com/asterisk/pjproject pjproject cd /usr/src/pjproject ./configure --prefix=/usr --enable-shared --disable-sound --disable-resample --disable-video --disable-opencore-amr ## um IPv6 Support in pjsip einzuschalten, muss das CFLAGS='-DPJ_HAS_IPV6=1' angegeben werden # IPV6 is turned off at default ! #./configure CFLAGS='-DPJ_HAS_IPV6=1' --prefix=/usr --enable-shared --disable-sound --disable-resample --disable-video --disable-opencore-amr # make dep make make install ldconfig ### check inst. # ldconfig -p | grep libpj ## System vorbereiten ## download Asterisk if [ ! -f /usr/src/$asteriskversion.tar.gz ] ; then wget http://downloads.asterisk.org/pub/telephony/asterisk/$asteriskversion.tar.gz fi if [ ! -d /usr/src/$asteriskversion ] ; then tar xvzf $asteriskversion.tar.gz fi ## erforderliche libs installieren /usr/src/$asteriskversion/contrib/scripts/install_prereq install ## optional /usr/src/$asteriskversion/contrib/scripts/get_mp3_source.sh /usr/src/$asteriskversion/contrib/scripts/get_ilbc_source.sh gcc -O2 /usr/src/$asteriskversion/contrib/utils/rawplayer.c -o /usr/bin/rawplayer ## asterisk installieren cd /usr/src/$asteriskversion ./configure make menuconfig make make install make samples make config make install-logrotate | | | -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] quickstart
I have the Asterisk book, it's enormous, the 4th edition as per http://www.asteriskdocs.org/. I'd like to do something like: http://www.voip-info.org/wiki/view/Asterisk+quickstart just to have two hardphones act as extensions and call each other. Is that a reasonable first task? I'm looking for the simplest litmus test for functionality possible. thanks, Thufir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
